August 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 24 Aug '15

24 Aug '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. No verification is made that the playback and capture codec setups are compatible in terms of number of TDM slots (where applicable). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V3: Updated commit message regarding the fact that the patch does not consider the slot setup in a TDM system with separate capture and playback codecs, after a comment from Pierre-Louis Bossart. Such a setup was not supported anyway before this patch, so let's handle that situation when and if it arises. V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v2 1/3] ALSA: hdac: Fix size allocation for ext device allocation
by Vinod Koul 23 Aug '15

23 Aug '15

While creating hdac_ext_device, we used hdev for sizeof insteadof edev, which resulted in eventual crash of the system Fix the size here Fixes: a512f5611646 ('ALSA: hdac: add hdac extended device') Reported-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 8544e4fb1cd3..4449d1a99089 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -139,7 +139,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) char name[15]; int ret; - edev = kzalloc(sizeof(*hdev), GFP_KERNEL); + edev = kzalloc(sizeof(*edev), GFP_KERNEL); if (!edev) return -ENOMEM; hdev = &edev->hdac; -- 1.9.1

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[alsa-devel] [PATCH 0/5] ALSA: hdac: More hdac core fixes and updates
by Vinod Koul 23 Aug '15

23 Aug '15

Hi Takashi, We had few more hdac core fixes & enhancements so senidng those. First one is fix for register pointers for spcap and link registers. This is required for SKL driver when we use link registers. rest add new APIs for powering down all links, getting stream from stream tag and direction, enabling SPIB. Last one remove keys as we dont need them for host streams Jeeja KP (5): ALSA: hdac: Fix to read the correct offset of spcap/link register ALSA: hdac: Add snd_hdac_ext_bus_link_power_down_all() ALSA: hdac: Add snd_hdac_get_hdac_stream() ALSA: hdac: Add support to enable SPIB for hdac ext stream ALSA: hdac: Remove the usage of key for host stream include/sound/hda_register.h | 4 +++ include/sound/hdaudio.h | 2 ++ include/sound/hdaudio_ext.h | 11 +++++++ sound/hda/ext/hdac_ext_controller.c | 30 +++++++++++++++--- sound/hda/ext/hdac_ext_stream.c | 62 ++++++++++++++++++++++++++++++++----- sound/hda/hdac_stream.c | 19 ++++++++++++ 6 files changed, 116 insertions(+), 12 deletions(-) Thanks -- ~Vinod

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[alsa-devel] [PATCH v2 0/3] ASoC: topology: update kcontrol ops binding & support TLV bytes controls
by mengdong.lin＠intel.com 22 Aug '15

22 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> History: v2: - let vendor specific kcontrol handlers override standard ones. - Merge handler binding for TLV bytes controls into soc_tplg_kcontrol_bind_io() Mengdong Lin (3): ASoC: topology: Bind vendor specific kcontrol handlers before standard ones ASoC: topology: Reduce arguments of soc_tplg_kcontrol_bind_io() ASoC: topology: Add support for TLV bytes controls include/sound/soc-topology.h | 13 +++++- sound/soc/soc-topology.c | 101 ++++++++++++++++++++++++++++--------------- 2 files changed, 79 insertions(+), 35 deletions(-) -- 1.9.1

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[alsa-devel] [patch] ALSA: hdsp: silence a sprinft() overflow warning
by Dan Carpenter 22 Aug '15

22 Aug '15

card->shortname is a 32 char string so the sprintf() can theoretically overflow. snd_rawmidi_new() can accept strings up to 64 bytes long. I have made the temporay buf[] array 40 bytes long and changed the sprintf() to snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..c1fe1d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1526,7 +1526,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) { - char buf[32]; + char buf[40]; hdsp->midi[id].id = id; hdsp->midi[id].rmidi = NULL; @@ -1537,7 +1537,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i hdsp->midi[id].pending = 0; spin_lock_init (&hdsp->midi[id].lock); - sprintf (buf, "%s MIDI %d", card->shortname, id+1); + snprintf(buf, sizeof(buf), "%s MIDI %d", card->shortname, id + 1); if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1;

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[alsa-devel] [PATCH v2] ALSA: ac97: Switch to dev_pm_ops
by Lars-Peter Clausen 21 Aug '15

21 Aug '15

Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single AC'97 driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. While we are at it also drop the ifdefs around the suspend/resume functions to increase compile test coverage. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch touches code in both ALSA and the input subsystem. Ideally this patch will be merged via the ALSA tree. Given that the wm97xx touchscreen driver is not seeing too many changes these days the risk of conflicts should be low. Changes since v1: * Drop ifdefs around suspend/resum functions --- drivers/input/touchscreen/wm97xx-core.c | 13 ++++--------- sound/ac97_bus.c | 26 -------------------------- 2 files changed, 4 insertions(+), 35 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index b1ae779..1534e9b 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -732,8 +732,7 @@ static int wm97xx_remove(struct device *dev) return 0; } -#ifdef CONFIG_PM -static int wm97xx_suspend(struct device *dev, pm_message_t state) +static int __maybe_unused wm97xx_suspend(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); u16 reg; @@ -765,7 +764,7 @@ static int wm97xx_suspend(struct device *dev, pm_message_t state) return 0; } -static int wm97xx_resume(struct device *dev) +static int __maybe_unused wm97xx_resume(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); @@ -799,10 +798,7 @@ static int wm97xx_resume(struct device *dev) return 0; } -#else -#define wm97xx_suspend NULL -#define wm97xx_resume NULL -#endif +static SIMPLE_DEV_PM_OPS(wm97xx_pm_ops, wm97xx_suspend, wm97xx_resume); /* * Machine specific operations @@ -836,8 +832,7 @@ static struct device_driver wm97xx_driver = { .owner = THIS_MODULE, .probe = wm97xx_probe, .remove = wm97xx_remove, - .suspend = wm97xx_suspend, - .resume = wm97xx_resume, + .pm = &wm97xx_pm_ops, }; static int __init wm97xx_init(void) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe..57a6dfc 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -27,35 +27,9 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return 1; } -#ifdef CONFIG_PM -static int ac97_bus_suspend(struct device *dev, pm_message_t state) -{ - int ret = 0; - - if (dev->driver && dev->driver->suspend) - ret = dev->driver->suspend(dev, state); - - return ret; -} - -static int ac97_bus_resume(struct device *dev) -{ - int ret = 0; - - if (dev->driver && dev->driver->resume) - ret = dev->driver->resume(dev); - - return ret; -} -#endif /* CONFIG_PM */ - struct bus_type ac97_bus_type = { .name = "ac97", .match = ac97_bus_match, -#ifdef CONFIG_PM - .suspend = ac97_bus_suspend, - .resume = ac97_bus_resume, -#endif /* CONFIG_PM */ }; static int __init ac97_bus_init(void) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: Set missing card owner field
by Axel Lin 21 Aug '15

21 Aug '15

Set the card owner field to prevent the module from being removed from underneath its users. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/blackfin/bfin-eval-adau1x61.c | 1 + sound/soc/intel/boards/byt-max98090.c | 1 + sound/soc/intel/boards/byt-rt5640.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 2 ++ sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 1 + sound/soc/rockchip/rockchip_max98090.c | 1 + sound/soc/rockchip/rockchip_rt5645.c | 1 + sound/soc/samsung/arndale_rt5631.c | 1 + sound/soc/samsung/snow.c | 1 + 13 files changed, 14 insertions(+) diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c index 4229f76..fddfe00c 100644 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ b/sound/soc/blackfin/bfin-eval-adau1x61.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { static struct snd_soc_card bfin_eval_adau1x61 = { .name = "bfin-eval-adau1x61", + .owner = THIS_MODULE, .driver_name = "eval-adau1x61", .dai_link = &bfin_eval_adau1x61_dai, .num_links = 1, diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9..d9f81b8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -126,6 +126,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { static struct snd_soc_card byt_max98090_card = { .name = "byt-max98090", + .owner = THIS_MODULE, .dai_link = byt_max98090_dais, .num_links = ARRAY_SIZE(byt_max98090_dais), .dapm_widgets = byt_max98090_widgets, diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b9..de9788a 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -197,6 +197,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { static struct snd_soc_card byt_rt5640_card = { .name = "byt-rt5640", + .owner = THIS_MODULE, .dai_link = byt_rt5640_dais, .num_links = ARRAY_SIZE(byt_rt5640_dais), .dapm_widgets = byt_rt5640_widgets, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59..c445312 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -185,6 +185,7 @@ static struct snd_soc_dai_link byt_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_byt = { .name = "baytrailcraudio", + .owner = THIS_MODULE, .dai_link = byt_dailink, .num_links = ARRAY_SIZE(byt_dailink), .dapm_widgets = byt_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 7810fc0..49f4869 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -275,6 +275,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "chtmax98090", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .aux_dev = &cht_max98090_headset_dev, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..7be8461 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -305,6 +305,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, @@ -317,6 +318,7 @@ static struct snd_soc_card snd_soc_card_chtrt5645 = { static struct snd_soc_card snd_soc_card_chtrt5650 = { .name = "chtrt5650", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..23fe040 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -323,6 +323,7 @@ static int cht_resume_post(struct snd_soc_card *card) /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 39e5660..684e8a7 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -136,6 +136,7 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { static struct snd_soc_card mt8173_max98090_card = { .name = "mt8173-max98090", + .owner = THIS_MODULE, .dai_link = mt8173_max98090_dais, .num_links = ARRAY_SIZE(mt8173_max98090_dais), .controls = mt8173_max98090_controls, diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index b722145..86cf975 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -191,6 +191,7 @@ static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = { static struct snd_soc_card mt8173_rt5650_rt5676_card = { .name = "mtk-rt5650-rt5676", + .owner = THIS_MODULE, .dai_link = mt8173_rt5650_rt5676_dais, .num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais), .codec_conf = mt8173_rt5650_rt5676_codec_conf, diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index acace20..cc26f81 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -147,6 +147,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "ROCKCHIP-I2S", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .aux_dev = &rk_98090_headset_dev, diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 3c6bb1e..0940279 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "I2S-RT5650", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .dapm_widgets = rk_dapm_widgets, diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 9e371eb..ee1fda9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -71,6 +71,7 @@ static struct snd_soc_dai_link arndale_rt5631_dai[] = { static struct snd_soc_card arndale_rt5631 = { .name = "Arndale RT5631", + .owner = THIS_MODULE, .dai_link = arndale_rt5631_dai, .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..07ce2cf 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -56,6 +56,7 @@ static int snow_late_probe(struct snd_soc_card *card) static struct snd_soc_card snow_snd = { .name = "Snow-I2S", + .owner = THIS_MODULE, .dai_link = snow_dai, .num_links = ARRAY_SIZE(snow_dai), -- 2.1.0

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Re: [alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 21 Aug '15

21 Aug '15

On Fri, 21 Aug 2015, Pierre-Louis Bossart wrote: > On 8/21/15 2:11 AM, Ricard Wanderlof wrote: > > > > On Thu, 20 Aug 2015, Pierre-Louis Bossart wrote: > > > >>> /** > >>> * snd_soc_runtime_activate() - Increment active count for PCM runtime components > >>> * @rtd: ASoC PCM runtime that is activated > >>> @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) > >>> > >>> /* first calculate min/max only for CODECs in the DAI link */ > >>> for (i = 0; i < rtd->num_codecs; i++) { > >>> + > >>> + /* > >>> + * Skip CODECs which don't support the current stream type. > >>> + * Otherwise, since the rate, channel, and format values will > >>> + * zero in that case, we would have no usable settings left, > >>> + * causing the resulting setup to fail. > >>> + * At least one CODEC should match, otherwise we should have > >>> + * bailed out on a higher level, since there would be no > >>> + * CODEC to support the transfer direction in that case. > >>> + */ > >>> + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], > >>> + substream->stream)) > >> > >> Maybe I misunderstood but shouldn't there be some sort of verification > >> that the codecs can use the same number of slots between playback and > >> capture if they share the same LRCLK/FS? e.g. it's not uncommon to have > >> 4 mic capture and 2 ch playback. If the capture and playback is handled > >> by different chips you'd still need to maintain some level of consistency. > > > > You're probably right, although it may be that that sort of thing is > > handled at a higher level, e.g. the machine driver in such a setup would > > configure both codecs to use TDM before we even get to this function, I > > don't know. > > > > I think we'd have to take care of that situation if and when it arises. > > The setup I have here doesn't allow me to test it. > > ok, sounds fine. maybe you could add this to the comments or commit > message then so that this assumption is known. Ok, good point. Will do. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs
by Takashi Iwai 21 Aug '15

21 Aug '15

Intel Haswell (and later) codec refreshes the widgets tree to expose the whole widget nodes at probing. However, this refresh was missing for sysfs tree. This patch adds the missing piece, as we have now a proper API. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a97db5fc8a15..df982fc6a872 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2331,6 +2331,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (is_haswell_plus(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + snd_hdac_refresh_widget_sysfs(&codec->core); } /* For Valleyview/Cherryview, only the display codec is in the display -- 2.5.0

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[alsa-devel] [patch] ALSA: hdsp: silence and underflow warning
by Dan Carpenter 21 Aug '15

21 Aug '15

I believe this probably cannot happen, as the code suggests. There would have to be an kcontrol->index.id which was zero, otherwise this would be prevented in snd_ctl_find_id(). But snd_BUG_ON() is just a WARN() or a no-op so static checkers complain that we keep on going with a negative offset. Let's just handle the error as well as printing a warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..468a95c 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2806,7 +2806,8 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); offset = ucontrol->id.index - 1; - snd_BUG_ON(offset < 0); + if (snd_BUG_ON(offset < 0)) + return -EINVAL; switch (hdsp->io_type) { case Digiface:

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